cf4ed1516e
GetRtpExtensions() is still used in one corner case for audio receive streams, so GetRtpExtensions has migrated to AudioReceiveStream. Updated FlexfecReceiveStream config management (incl. pass by value) and now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl. Call GetRtpExtensionMap() from call.cc instead of constructing one on the fly for each rtp packet (for video packets at least). Bug: webrtc:11993 Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36839}
100 lines
3.6 KiB
C++
100 lines
3.6 KiB
C++
/*
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* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_RECEIVE_STREAM_H_
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#define CALL_RECEIVE_STREAM_H_
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#include <vector>
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#include "api/crypto/frame_decryptor_interface.h"
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#include "api/frame_transformer_interface.h"
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#include "api/media_types.h"
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#include "api/scoped_refptr.h"
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#include "api/transport/rtp/rtp_source.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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namespace webrtc {
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// Common base interface for MediaReceiveStream based classes and
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// FlexfecReceiveStream.
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class ReceiveStream {
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public:
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// Receive-stream specific RTP settings.
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// TODO(tommi): This struct isn't needed at this level anymore. Move it closer
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// to where it's used.
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struct ReceiveStreamRtpConfig {
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// Synchronization source (stream identifier) to be received.
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// This member will not change mid-stream and can be assumed to be const
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// post initialization.
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uint32_t remote_ssrc = 0;
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// Sender SSRC used for sending RTCP (such as receiver reports).
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// This value may change mid-stream and must be done on the same thread
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// that the value is read on (i.e. packet delivery).
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uint32_t local_ssrc = 0;
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// Enable feedback for send side bandwidth estimation.
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// See
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// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
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// for details.
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// This value may change mid-stream and must be done on the same thread
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// that the value is read on (i.e. packet delivery).
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bool transport_cc = false;
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// RTP header extensions used for the received stream.
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// This value may change mid-stream and must be done on the same thread
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// that the value is read on (i.e. packet delivery).
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std::vector<RtpExtension> extensions;
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};
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// Set/change the rtp header extensions. Must be called on the packet
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// delivery thread.
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virtual void SetRtpExtensions(std::vector<RtpExtension> extensions) = 0;
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virtual RtpHeaderExtensionMap GetRtpExtensionMap() const = 0;
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// Returns a bool for whether feedback for send side bandwidth estimation is
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// enabled. See
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// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
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// for details.
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// This value may change mid-stream and must be done on the same thread
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// that the value is read on (i.e. packet delivery).
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virtual bool transport_cc() const = 0;
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protected:
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virtual ~ReceiveStream() {}
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};
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// Either an audio or video receive stream.
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class MediaReceiveStream : public ReceiveStream {
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public:
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// Starts stream activity.
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// When a stream is active, it can receive, process and deliver packets.
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virtual void Start() = 0;
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// Stops stream activity. Must be called to match with a previous call to
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// `Start()`. When a stream has been stopped, it won't receive, decode,
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// process or deliver packets to downstream objects such as callback pointers
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// set in the config struct.
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virtual void Stop() = 0;
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virtual void SetDepacketizerToDecoderFrameTransformer(
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rtc::scoped_refptr<webrtc::FrameTransformerInterface>
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frame_transformer) = 0;
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virtual void SetFrameDecryptor(
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rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
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virtual std::vector<RtpSource> GetSources() const = 0;
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};
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} // namespace webrtc
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#endif // CALL_RECEIVE_STREAM_H_
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